The present invention relates to systems which use multiple microphones to reduce the noise and to enhance a target signal.
Such systems are called beamforming systems or directional systems. FIG. 1 shows a simple two-microphone system that uses a fixed delay to produce a directional output. The first microphone 22 is separated from the second microphone 24 by distance. The output of the second microphone 24 is sent to a constant delay 26. In one case, a constant delay, d/c where c is the speed of sound, is used. The output of the delay is subtracted from the output of the first microphone 22. FIG. 1B is a polar pattern of the gain of the system of FIG. 1A. The delay d/c causes a null for signals coming from the 180xc2x0 direction. Different fixed delays produce polar patterns having nulls at different angles. Note that at the zero degree direction, there is very little attenuation. The fixed directional system of FIG. 1A is effective for the case that the target signal comes from the front and the noise comes exactly from the rear, which is not always true.
If the noise is moving or time-varying, an adaptive directionality noise reduction system is highly desirable so that the system can track the moving or varying noise source. Otherwise, the noise reduction performance of the system can be greatly degraded.
FIG. 2 is a diagram in which the output of the system is used to control a variable delay to move the null of the directional microphone to match the noise source.
The noise reduction performance of beamforming systems greatly depends upon the number of microphones and the separation of these microphones. In some application fields, such as hearing aids, the number of microphones and distance of the microphones are strictly limited. For example, behind-the-ear hearing aids can typically use only two microphones, and the distance between these two microphones is limited to about 10 mm. In these cases, most of the available algorithms deliver a degraded noise-reduction performance. Moreover, it is difficult to implement, in real time, such available algorithms in this application field because of the limits of hardware size, computational speed, mismatch of microphones, power supply, and other practical factors. These problems prevent available algorithms, such as the closed-loop-adapted delay of FIG. 2, from being implemented for behind-the-ear hearing aids.
It is desired to have a more practical system for implementing an adaptive directional noise reduction system.
The present invention is a system in which the outputs of the first and second microphones are sampled and a discrete Fourier Transform is done on each of the sampled time domain signals. A further processing step takes the output of the discrete Fourier Transform and processes it to produce a noise canceled frequency-domain signal. The noise canceled frequency-domain signal is sent to the Inverse Discrete Fourier Transform to produce a noise canceled time domain data.
In one embodiment of the present invention, the noise canceled frequency-domain data is a function of the first and second frequency domain data that effectively cancels noise when the noise is greater than the signal and the noise and signal are not in the same direction from the apparatus. The function provides the adaptive directionality to cancel the noise.
In another embodiment of the present invention, the function is such that if X(xcfx89) represents one of the first and second digital frequency-domain data and Y(xcfx89) represents the other of the first and second digital frequency-domain data, the function is proportional to X(xcfx89)[1xe2x88x92|Y(xcfx89)|X(xcfx89)|].
The present invention operates by assuming that for systems in which the noise is greater than the signal, the phase of the output of one of the Discrete Fourier Transforms can be assumed to be the phase of the noise. With this assumption, and the assumption that the noise and the signal come from two different directions, an output function which effectively cancels the noise signal can be produced.
In an alternate embodiment of the present invention, the system includes a speech signal pause detector which detects pauses in the received speech signal. The signal during the detected pauses can be used to implement the present invention in higher signal-to-noise environments since, during the speech pauses, the noise will overwhelm the signal, and the detected xe2x80x9cnoise phasexe2x80x9d during the pauses can be assumed to remain unchanged during the non-pause portions of the speech.
One objective of the present invention is to provide an effective and realizable adaptive directionality system which overcomes the problems of prior directional noise reduction systems. Key features of the system include a simple and realizable implementation structure on the basis of FFT; the elimination of an additional delay processing unit for endfire orientation microphones; an effective solution of microphone mismatch problems; the elimination of the assumption that the target signal must be exactly straight ahead, that is, the target signal source and the noise source can be located anywhere as long as they are not located in the same direction; and no specific requirement for the geometric structure and the distance of these dual microphones. With these features, this scheme provides a new tool to implement adaptive directionality in related application fields.